PBX Service Pack: 22.214.171.124-2
I’m seeing an issue where remote telephones get one-way audio; the remote phone is sending audio to the PBX on the wrong RTP port.
NOTE: Yes, I have the correct settings for sip_nat.conf via the “Astersik SIP Settings” tab in FreePBX
I’ve used Wireshark to verify the PBX sends a correct RTP port number to the endpoint in the SIP message (e.g. 10998), however the endpoint seems to ignore or mis-translate and send audio on a totally different port (e.g. 10672).
The strange thing is that these remote endpoints (a Polycom and a SipDroid phone) were working on the PBX as a TrixboxCE system but seemed to break when the PBX was upgraded to FreePBX distro. The routers in place haven’t changed (Cisco RVS4000). SIP ALG is not turned on in either router, 5060 & RTP ports are forwarded on the PBX router.
Even stranger yet: this issue only occurs when a call is placed on a SIP trunk - extension to extension calls seem just fine. BTW: Re-invite is turned off on the trunk.
Does this ring any bells?